Outgoing Call No Voice On Both Sides/Directions

[Hary_CsC]

Hi Guys.

I have the problem of no voice in both way when Call Outgoing.

(IP Phone)7317356 -> (CUCM)10.10.40.200 -> (Cube)10.10.255.10 -> (CE)10.21.10.234 -> (PE)10.21.10.234 -> (Destination) 087888555213

*attached VG debug

Thank You

[b.winter]

Have you checked your own logs?

If yes, then you would see, that the call is not even successfully established.

Your CUBE sends multiple "200 OK" back to CUCM, but CUCM isn't reacting to that (an ACK message is missing). Because of this, CUBE tries to hang up the call with a "BYE", but also these "BYE" messages aren't answered by CUCM.

Something is wrong with your connection between CUCM and CUBE. You should check that.

Also: Configure the dial-peers / tenants you use for the connection between CUBE and CUCM as TCP. That's more reliable.

It could also be, that the "200 OK" and the "BYE" messages are source from the wrong interface and then it's clear that CUCM doesn't react.

[Hary_CsC]

Hai @b.winter 

Have a nice day.

It's true what you said there was an error on the IP interface given to cucm. (c=IN IP4 x.x.x.x )

Now the outgoing call is successful.But the Incoming Call rings but can't be picked up.

Thank You

[b.winter · Accepted solution]

Hi,

• Why do your "internal" dial-peers 1 and 2 use Loopback0 as signalling/media interface? It is described as management-interface in your config. Shouldn't it be Port-channel1 (to which IP-address does the SIP trunk in CUCM point to?)?
• Dial-peer 101: Re-Add the signalling/media interface bind command with the interface gig0/0/2
• Dial-peer 101: Delete command session-target (it's no used in incoming dial-peers).
• Dial-peer 101 and 201: Why do you use the command "voice-class sip transport switch udp tcp"?
• Dial-peer 101 and 201: Why do have a codec list in dial-peer 101 configured, but in 201 you have set it to a fixed codec?
• Where is your dial-peer "from CUCM to VG"?

I also would set up the whole dial-peer matching completely differently, just using the knowledge I have from your config / debug:

voice translation-profile TO_CUCM --> new profile
translate called 3
!
dial-peer voice 1 voip
description *** From VG to CUCM-PUB ***
destination-pattern 0212356 --> change number
translation-profile outgoing TO_CUCM --> add command
session protocol sipv2
session target ipv4:10.10.40.200
voice-class codec 1
no voice-class sip transport switch udp tcp --> delete command
voice-class sip early-offer forced
voice-class sip bind control source-interface Port-channel1 --> change interface
voice-class sip bind media source-interface Port-channel1 --> change interface
dtmf-relay rtp-nte
no vad
!
dial-peer voice 2 voip
description *** From VG to CUCM-SUB ***
destination-pattern 0212356 --> change number
translation-profile outgoing TO_CUCM --> add command
session protocol sipv2
session target ipv4:10.10.40.201
voice-class codec 1
no voice-class sip transport switch udp tcp --> delete command
voice-class sip early-offer forced
voice-class sip bind control source-interface Port-channel1 --> change interface
voice-class sip bind media source-interface Port-channel1 --> change interface
dtmf-relay rtp-nte
no vad
!
dial-peer voice 2 voip --> new dial-peer
description *** From CUCM-to VG ***
incoming called-number 9T
session protocol sipv2
voice-class codec 1
voice-class sip bind control source-interface Port-channel1
voice-class sip bind media source-interface Port-channel1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 101 voip
description *** Incoming via SIP-JAYAKOM ***
no translation-profile incoming JAYAKOM-INCOMING-1 --> delete the profile here
session protocol sipv2
no session target ipv4:10.21.10.233 --> delete command
incoming called-number 0212356 --> Match based on your DID range (and not on all with a dot ".")
voice-class codec 1
no voice-class sip transport switch udp tcp --> delete command
voice-class sip early-offer forced
voice-class sip bind control source-interface GigabitEthernet0/0/2 --> add command
voice-class sip bind media source-interface GigabitEthernet0/0/2 --> add command
dtmf-relay rtp-nte
no vad
!
dial-peer voice 201 voip
description *** Outgoing via SIP-JAYAKOM ***
translation-profile outgoing JAYAKOM_OUTGOING
preference 1
destination-pattern 9T
session protocol sipv2
session target ipv4:10.21.10.233
no voice-class sip transport switch udp tcp --> delete command
voice-class sip early-offer forced
voice-class sip bind control source-interface GigabitEthernet0/0/2
voice-class sip bind media source-interface GigabitEthernet0/0/2
dtmf-relay rtp-nte
no codec g711alaw --> delete command
voice-class codec 1 --> add command
no vad